1. Field of the Invention
This invention relates generally to improving the quality of sound generated by one or more low-frequency loudspeakers in a listening area, and more particularly, to reduced latency design of digital filters.
2. Related Art
Sound systems typically include loudspeakers that transform electrical signals into acoustic signals. The loudspeakers may include one or more transducers that produce a range of acoustic signals, such as high, mid and low-frequency signals. An example of such a loudspeaker is a subwoofer that may include a low frequency transducer that typically produces low-frequency signals in the range of 20 Hz to 100 Hz.
Sound systems may generate the acoustic signals in a variety of listening environments, such as, home listening rooms, home theaters, movie theaters, concert halls, vehicle interiors, and recording studios, to name a few. Typically, a listening environment includes single or multiple listening positions for person or persons to hear the acoustic signals generated by the loudspeakers. The listening position may be a seated position, such as a section of a couch in a home theater environment, or a standing position, such as a spot where a conductor may stand in a concert hall.
The listening environment may affect the acoustic signals, including the low, mid and high frequency signals at the listening positions. Depending on the acoustic characteristics of the room, the position of the listener in a room and the position of the loudspeaker in the room, the loudness of the sound can vary for different frequencies. This may be especially true for low frequencies.
Low frequencies may be important to the enjoyment of music, movies, and other forms of audio entertainment. In the home theater example, the room boundaries, including the walls, draperies, furniture, furnishings, and the like, may affect the acoustic signals as they travel from the loudspeakers to the listening positions.
One approach to characterizing the room is by the impulse response of a loudspeaker received at a microphone placed within the listening area. The impulse response contains information about various properties of the acoustical signals including the amplitude and phase at a single frequency, a discrete number of frequencies, or a range of frequencies.
An amplitude response approach is a measurement of the loudness at the frequencies of interest. Generally, the amplitude is measured in decibels (dB). Amplitude deviations may be expressed as positive or negative decibel values in relation to a designated target value. The closer the amplitude values measured at a listening position are to the target values, the better the amplitude response. Deviations from the target reflect changes that occur in the acoustic signal as it interacts with room boundaries. Peaks represent a positive amplitude deviation from the target, while dips represent a negative amplitude deviation from the target.
Deviations in amplitude may depend on the frequency response of the acoustic signal reproduced at the subwoofer, the subwoofer location, and/or the listener position. A listener may not hear low frequencies as they were recorded on the recording medium, such as a soundtrack or compact disk, but instead as they are distorted by the room boundaries. Thus, the room can change the acoustic signal reproduced by the subwoofer and adversely affect the low-frequency performance of the sound system.
Many equalization techniques have been used in the past to reduce or remove amplitude deviations within a listening area. One such technique is spatial averaging, which calculates an average amplitude response for multiple listening positions, and then equally implements the equalization for all subwoofers in the system. Spatial averaging, however, only corrects for a single “average listening position” that does not exist in reality. Thus, when using spatial averaging techniques, some listening positions may have a better low-frequency performance than other positions, but certain locations may be negatively impacted. For instance, the spatial averaging may worsen the performance at some listening positions as compared to their un-equalized performance. Moreover, attempting to equalize and flatten the amplitude response for a single location potentially creates problems. While peaks may be reduced at the average listening position, attempting to amplify frequencies where dips occur requires significant additional acoustic output from the subwoofer, thus reducing the maximum acoustic output of the system and potentially creating large peaks in other areas of the room.
Another known equalization technique is to position multiple subwoofers in a “mode canceling” arrangement. By locating multiple loudspeakers symmetrically within the listening room, standing waves may be reduced by exploiting destructive and constructive interference. However, the symmetric “mode canceling” configuration assumes an idealized room. (i.e., dimensionally and acoustically symmetric) and does not account for actual room characteristics including variations in shape or furnishings. Moreover, the symmetric positioning of the loudspeakers may not be a realistic or desirable configuration for the particular room setting.
Still another equalization technique is to configure the audio system using mathematical analysis to reduce amplitude deviations. One such mathematical analysis simulates standing waves in a room based on room data. For example, room dimensions, such as length, width, and height of a room, are input and various algorithms predict where to locate a subwoofer based on data input. This mathematical method does not, however, account for the acoustical properties of a room's furniture, furnishings, composition, etc. For example, an interior wall having a masonry exterior may behave very differently in an acoustic sense than a wood framed wall. Further, this mathematical method cannot effectively compensate for partially enclosed rooms and may become computationally onerous if the room is not rectangular.
Another method to equalize the frequency responses in a room attempts to reduce variations across seats by optimizing a limited set of parameters such as gain, delay and a set of low order filter parameters for each transducer using trial and error. However, the accuracy of that method is more by chance because of the guessing involved in determining those parameters. As such, to obtain an accurate equalization solution, it takes a tremendous amount of computational power for a full search. Moreover, the method does not in general yield optimum results because of the limited complexity of the applied filters, which is necessary to limit the processing time.
Accordingly, a need exists for a system that is able to accurately determine a configuration for an audio system such that the audio performance for one or more listening positions in a given space is improved with minimum latency.